Method and apparatus for mode balance for analog FM, digital, radio blend logic in an automotive environment

ABSTRACT

A radio includes a first tuner and a second tuner. A processor compares a first perceivable volume level of a station tuned by the first tuner to at least one second perceivable volume level of at least one background station tuned by the second tuner. The processor enables automatic volume knob changes using a pre-calibrated lookup table that associates a volume step of the volume knob with a difference between the first perceivable volume level and the second perceivable volume level.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a continuation of U.S. patent application Ser. No.14/529,954, filed on Oct. 31, 2014, now U.S. Pat. No. 9,559,657, issuedon Jan. 31, 2017, and claims benefit to U.S. Provisional PatentApplication Ser. No. 61/898,206, filed on Oct. 31, 2013, which areincorporated herein by reference in their entirety.

BACKGROUND OF THE INVENTION Description of the Prior Art

Mode balancing is a technique used in radio head units to provideconstant perceived volume levels when the user traverses across thedifferent mode sources such as CD, Multimedia, AUX, FM and/or AM. Audioattenuation levels are placed in a way to make the volume perceptuallythe same and prevent customer listening discomfort. Without modebalance, the user would experience a sudden discomfort due to perceivedvolume increase for the same volume step he has set on the radio. Whilethe concept of mode balance applies to different sources, achieving modebalance within a source is difficult to achieve. This in turn applies tomultimedia content that can be stored with different compression rates.

The interplay between Analog, Digital and IP Radio is now coming tofruition due to connectivity availability in the car radio head unit,either through tethering of a cell phone or a dedicated embedded cellmodem. FIG. 1 illustrates the current interplays and use cases that nowexist due to known enabling technologies. The interplay on the IP frontcan take on facets that encompass mobile TV use cases depending on howthe broadcaster scope and scale takes shape. The current state of artused in car radio head units is the following:

When a user changes from one source to another, the current state of artemploys timed fade-in and fade-out mechanisms. As an example, switchingfrom an AM source to an FM source uses a 500 ms fade-in (the currentaudio source is slowly muted with audio ramp) and fade-out (the newaudio source is slowly unmuted with audio ramp). This approach does notscale for within a source.

Digital radio and improved connectivity bring about new permutations forthe end user for continued audio listenership of the same audio viadifferent broadcast media (e.g., IP Radio, Analog AM/FM, AM HD IBOC, FMHD IBOC and SiriusXM). Connectivity and new bridging enablers likeRadioDNS (www.radiodns.org) allow the user to link to IP radio. A usercan be listening to an AM/FM station and if the signal fades he cancontinue to listen to the current song through either the Digital Radiotransition or IP Radio through enabling technologies such as FM to DABlink or Radio DNS. FM to DAB link applies in Europe where the Programidentification (PI) code of a radio station is linked to the DABstation's Service ID that serves as basis of confirmation of the sameaudio content to allow for a switch. Radio DNS applies to embedded cellmodems and involves an off board server keeping a table that maps the PIcode of a radio station to the URL address of the station's InternetProtocol station. An example for Radio DNS is in the USA where the radiostation span is within the range of the transmitter. The concept ofcontinued reception is now a possibility with improved connectivity. TheIP switch typically will be based on a cost based optimization for theend user, meaning a switch from “free” analog or HD IBOC AM/FM stations(where accessible) to IP radio based on a model of the connectivitysubscription model used. A problem is that changing between differentbroadcast media for the same simulcast listened-to broadcast audio canbring about a difference in perceived volume levels. The RadioDNSmapping can allow the radio head unit to trigger the switch to IP Radioif the current tuned signal medium fades away or vice versa, providedthe radio head unit includes a browser.

Consider the following use cases and current issue illustrated in theuse case table of FIGS. 2A-B. At a high level Use Cases 5 and 9 can begrouped as examples of intra-station effects with the user experiencinga difference in perceived audio volume by simply staying tuned to thecurrent station in the presence of audio content modulation or bandwidthchanges due to program content. The other use cases can be groupedtogether as inter (between) station effects.

In case of Use case 5, the present state of art protects only thetransition from analog to digital HD IBOC and vice versa by differentmethods. This includes slowly increasing the bandwidth from analog todigital. The current state of art however does not address the temporallong term effect to the driver. Field test experience has shown that theAM to AM HD transition can have a dramatic increase in volume levelwhich causes a typical end user to manually increase or decrease thevolume level depending on the transition.

From the end user experience, the end user typically chooses to manuallychange the volume levels to a level that is more pleasing to the ear forlistener comfort. Hereinbelow is an explanation of the science behindthe issues noted from the end user perspective:

-   1) Explanation of the meaning of bandwidth and its effect on    perceived volume and explanation of the scientific basis behind    this.-   2) Explanation of the use case of how mode balance within a source,    such as FM, is not supported by current radio head units in the    market space.    Bandwidth and Perceived Volume

Audio files with the same amplitude level and same volume settings butwith differing bandwidth levels may have differing levels of perceivedloudness or perceived volume. The perceived loudness difference isattributed to the human auditory system which acts as a dedicated bandpass filter for all frequencies that can be detected by the human ear.Within each critical band, perceptual loudness is dominated by thefrequencies with the strongest intensities. The underlying principle isthat when energy within a band is fixed, the loudness remains constant.However, once the bandwidth is exceeded, meaning that the energy isspread over more than one critical hand, there is an increase inperceived loudness.

Although two audio signals may have the sample amplitude levels, thesignal with the higher sampling rate may sound perceivably louder thanthe one with the lower sampling rate. The scientific basis behind thisis described hereinbelow. The number of equivalent rectangular bandwidth(ERB) auditory/critical filters as a function of frequency F is givenby:P(F)=21.4 log₁₀(4.37F+1)  (1)

Accordingly, if there is a 15 kHz sampling rate, setting F to 7.5 kHzresults in thirty-two auditory filters. The total loudness metric asapplied to demodulated FM audio for a particular volume step is definedas:

$\begin{matrix}{{L = {\int_{0}^{P}{{L(p)}{dp}}}},{{{where}\mspace{14mu} p} = {32\mspace{14mu}{for}\mspace{14mu}{{FM}.}}}} & (2)\end{matrix}$

The metric defined in (2) may be equivalent to the physiologicalresponse of the total neural activity evoked. The above principles maybe important in determining the volume levels produced by a radio headunit as perceived by a human. Hereinbelow, these principles are appliedto real life use examples daunting the radio head unit end user.

The current FM station user experience is an example of how achievingmode balance within a source is a difficult value proposition atpresent. Stations may operate at modulation levels up to 75 kHz for theNorth America market where stations are spaced out at 200 kHz steps andtypically have bandwidths of 22.5 to 40 kHz. In Europe and Rest OfWorld, stations are spaced out at 100 kHz steps.

Taking North America as an example, for a fixed volume setting, stationswhich broadcast talk shows at modulation frequencies around 88.1 MHzsound ‘soft’ when they are tuned to, while other stations whichbroadcast talk shows around 95.5 MHz or 90.1 MHz sound ‘louder’. Thatis, the reason for differing loudness levels is the fact that stationsoperate at different modulation levels.

FM modulation, the levels of which are related to broadcast bandwidth,is a form of angle modulation in which the base-band signal modulatesthe frequency of a carrier wave. In FM modulation theory, theinstantaneous frequency of an FM modulation station is depicted by thefollowing formula:F _(i)(t)=F _(c*)(t)+K _(VCO) *m(t)  (3)where F_(i)(t) is the instantaneous FM frequency deviation, F_(c) is thecarrier frequency, K_(VCO) is the voltage to frequency gain of theVoltage Controlled Oscillator (VCO) with units of Hz/V and m(t) is themessage signal.

The instantaneous phase of the output FM signal in turn translates to:

$\begin{matrix}{\Theta = {{{2\pi\; F_{c}} \star (t)} + {2\pi\;{Kvco}{\int_{0}^{t}{{m(t)}{dt}}}}}} & (4)\end{matrix}$where Θ is the instantaneous phase of the transmitter signal, F_(c) isthe carrier frequency, K_(VCO) is the voltage to frequency gain of theVoltage Controlled Oscillator (VCO) with units of Hz/V and m(t) is themessage signal that is modulated onto the carrier wave. The functionm(t) may be either speech or music that is transmitted by the FM radiostations, and the typical bandwidth allowed for these two entities is 15kHz. The carrier wave frequency ranges from 87.7 to 107.9 MHz for theNorth American market and 87.5 to 108.0 MHz for the European and Rest OfWorld market.

As such, the transmitted FM signal is defined as:

$\begin{matrix}{{X_{FM}(t)} = {A_{C}{\cos\left\lbrack {{{2\pi\; F_{c}} \star (t)} + {2\pi\;{Kvco}{\int_{0}^{t}{{m(t)}{dt}}}}} \right\rbrack}}} & (5)\end{matrix}$where F_(c) is the carrier frequency, K_(VCO) is the voltage tofrequency gain of the Voltage Controlled. Oscillator (VCO) with units ofHz/V and m(t) is the message signal that is modulated onto the carrierwave and A_(C) is the amplitude of the FM carrier signal.

As noted by Equation 5, the amplitude of the FM signal is constantregardless of the modulated message signal. If m(t) is set to be A_(m)cos 2π F_(m)(t) for example, Equation 5 can be rewritten as:

$\begin{matrix}{{X_{FM}(t)} = {A_{C}{\cos\left\lbrack {{{2\pi\; F_{c}} \star (t)} + {\frac{{Kvco} \star {Am}}{Fm}\sin\; 2\pi\;{F_{m}(t)}}} \right\rbrack}}} & (6)\end{matrix}$

In Equation 6 the value K_(VCO)*Am can be termed into Δf, therebytranslating Equation 6 into the following:

$\begin{matrix}{{X_{FM}(t)} = {A_{C}{\cos\left\lbrack {{{2\pi\; F_{c}} \star (t)} + {\frac{\Delta\; f}{Fm}\sin\; 2\pi\;{F_{m}(t)}}} \right\rbrack}}} & (7)\end{matrix}$

In North America, Δf translates to a maximum bound of 75 kHz due to the200 kHz FM step size, while in Europe and Rest Of the World whichutilizes 100 kHz FM frequency step size, Δf translates to a maximumhound of slightly over 22.5 kHz. Fm is typically about 15 kHz for FMstations.

This mathematical exercise leads to the point that Δf is the peakfrequency deviation of the FM signal from the center frequency of theCarrier Wave and is directly proportional to the amplitude of themodulated message signal (Am) and the gain of the VCO (K_(VCO)).

From a typical end user standpoint, the end result of different radiostation frequencies having differing loudness levels is that the driverends up manually turning up the volume knob of the radio head unit tohear better (e.g., to make the sound more discernible) for a lowmodulation station. In contrast, when a user tunes to a station with ahigher modulation levels, the user may end up manually turning down thevolume level. This is a real life cumbersome problem being faced by endusers at the present.

In low end radios in the USA, the operation of seek typically causes anaudio mute for the duration of the time period in which the radio seeksthrough the band to find the next strong station. After the next strongstation is found, the station is tuned to and un-muting is performed. Inmid-range radio head units used in the Europe and North America markets,the use of dual tuners radios is slowly becoming main stream. Theintroduction of the second tuner supports scaling for phase diversityand building up of the station list when the second tuner performsbackground scanning of the FM band without the user being aware of it.As improvements in the technology are made, the seek operation isbecoming nearly instantaneous and the issue of trying to resolve the‘increased volume levels’ during transition from low modulation to highmodulation stations is becoming more important.

The current state of art for HD Radio ensures that the analog to digitaltransition and digital to analog transition sounds smooth to the userbut does not provide constant mode balance to the end user. That is,bandwidths are slowly increased and decreased in a manner to ensure thatthe end user does not hear the perceived audio difference. But if atemporal time perspective is taken at the end, then the user ends uphearing a difference in volume levels.

With digital radio such as HD (In Band On Channel), if an FM stationthat is tuned to happens to be an analog station, and a digital stationthat is being switched to has a strong signal, then typically after fiveseconds the radio receives digital audio synchronization and shifts toHD reception. Consequently, the perceived audio levels increase as theradio receiver transitions from the tuned analog station to the digitalradio. The reason for this is that an analog FM station operates at 15kHz while an HD FM station operates up to 20 kHz. Likewise, when theprimary HD station experiences heavy multipath or starts fading, HD FMBlend occurs whereby the radio receiver end user perceives a differencein audio levels as the signal transitions from digital (up to 20 kHz) toanalog (up to 15 kHz).

The table in FIG. 3 shows data relevant to perceived audio levels.

SUMMARY OF THE INVENTION

The present invention may be applied to HD Digital Radio applications toaddress the above-described problems. The invention may provide a methodand system for achieving audio mode balance within and across sources.More particularly, the invention may provide a solution to the problemthat changing between different broadcast media for the same simulcastlistened-to broadcast audio can bring about a difference in perceivedvolume levels. The novel method of the invention may adjust volumedynamically and may provide a perceived volume metric and avoiddiscomfort to the end user. The inventive method may take into accountthe car cabin environment which has engine noise and road noise, both ofwhich can affect perceived volume levels. A microphone input can be usedaccordingly.

The present invention may automatically increase or decrease the volumelevel for the end user by employing novel techniques that address theuse cases noted in the table of FIGS. 2A-B. Depending on the mediumused, the increase or decrease in volume level can be dealt with byeither using bandwidth extension algorithms, or automatically increasingthe volume level for the end user through a trigger from the Radio HUsoftware.

Bandwidth extension algorithms are typically better suited for digitalaudio sources like DAB, HD IBOC and other multimedia digital audiosources. Bandwidth extension algorithms may not work well for analogsignals which are prone to noise levels triggered by multipath, adjacentchannel or frequency offset errors which can cause an increase in noiselevels. Bandwidth extension may apply only when the quality of theanalog signal is deemed to be good for the long term. The bandwidthextension comes into play in such a manner so as to normalize alldigital audio sources to the same bandwidth using techniques likespectral band replication, etc.

In bandwidth normalization, if the user is listening to a low bandwidthstation and all neighboring stations are higher bandwidth, bandwidthextension may be applied to the station using information from theneighboring stations' audio bandwidth. Considering use cases 7 and 8from the table of FIGS. 2A-B, the bandwidth normalization may bepossible only if the radio head unit can gauge the currently tunedstation's audio bandwidth as compared to the bandwidth of theneighboring stations that are potential targets that the end user canswitch to.

FIG. 4 illustrates the audio channel path of the present invention,which can reservable typical flow within a System On Chip (SOC) or ASICso background audio decode may be employed by multituners and/or asingle tuner with wideband ADC and background audio decode capability.For the main audio entertainment path, bandwidth extension applies todigital audio. However, volume detection applies to analog audio. Forthe second audio path of the background tuner, volume detection alsoapplies.

The volume level for the end user may be automatically increased througha trigger from the radio head unit software which may use the volumeloudness level as a metric. This may be better suited for pure analogstations with different modulation levels. The volume increment ordecrement steps factor in the Munson Curve to ensure volume gains areapplied in such a manner to protect the psychoacoustic model. The Munsonis in essence an inversion of the psychoacoustic model such that itprovides additional gain to frequencies which the ear is inherently hardof hearing. A calibratable table can be used to translate difference inperceived volume levels to actual volume level step increase or decreaseso as to maintain a same mode balance across analog stations.

FIG. 9 is an Equal Loudness Curve which highlights the intensity levelsof frequency components required for the human ear to perceive it asequal volume for a particular volume step. The munson curve compensationprovided by DSP works under the premise that the input signal containsthe full spectrum. If, however, the broadcast station curtails the highbandwidth, then even if the radio receiver supports the munsoncompensation, then there may be no optimum perceived impact.

In one embodiment, the invention comprises a radio including a firsttuner and a second tuner. A processor compares a first perceivablevolume level of a station tuned by the first tuner to at least onesecond perceivable volume level of at least one background station tunedby the second tuner. The processor enables automatic volume knob changesusing a pre-calibrated lookup table that associates a volume step of thevolume knob with a difference between the first perceivable volume leveland the second perceivable volume level.

In another embodiment, the invention comprises a radio including a firsttuner and a second tuner. A processor compares a first perceivablevolume level of a station tuned by the first tuner to at least onesecond perceivable volume level of at least one background station tunedby the second tuner. A volume change is performed dependent upon adifference between the first perceivable volume level and the secondperceivable volume level.

In yet another embodiment, the invention includes a radio including afirst tuner and a second tuner. A processor compares a first perceivablevolume level of a station tuned by the first tuner to at least onesecond perceivable volume level of at least one background station tunedby the second tuner. The processor enables automatic volume knob changesdependent upon a volume step of the volume knob and a difference betweenthe first perceivable volume level and the second perceivable volumelevel.

BRIEF DESCRIPTION OF THE DRAWINGS

The above-mentioned and other features and objects of this invention,and the manner of attaining them, will become more apparent and theinvention itself will be better understood by reference to the followingdescription of embodiments of the invention taken in conjunction withthe accompanying drawings, wherein:

FIG. 1 is a flow chart illustrating known technologies.

FIG. 2A is a first portion of a table illustrating processes of theprior art.

FIG. 2B is a second portion of the table of FIG. 2A.

FIG. 2C is a third portion of the table of FIG. 2A.

FIG. 3 is a table illustrating service and related signal qualitymetrics of the prior art.

FIG. 4 illustrates an Audio Channel Path of the present invention.

FIG. 5A is a first portion of a table illustrating some cases to whichthe present invention may be applied.

FIG. 5B is a second portion of the table of FIG. 5A.

FIG. 6 shows a 512-sample audio signal frame normalized spectrum plot.

FIG. 7 is a table mapping out the appropriate volume level settings thatmay be applied to ensure that an end user perceives a specified volumelevel in a digital use case.

FIG. 8 is a table mapping out the appropriate volume level settings thatmay be applied to ensure that an end user perceives a specified volumelevel in an analog use case.

FIG. 9 is a Munson curve.

DETAILED DESCRIPTION

The embodiments hereinafter disclosed are not intended to be exhaustiveor limit the invention to the precise forms disclosed in the followingdescription. Rather the embodiments are chosen and described so thatothers skilled in the art may utilize its teachings.

The main audio entertainment path may apply for the currently tunedaudio source. The audio therein may be routed to a core which can runthe volume detection algorithm. The second audio path which can applyfor audio may be decoded/demodulated by background tuners and orwideband front end devices for other than the currently tuned stationaudio, and may be routed to volume detection to gauge the volume levelsof the currently tuned audio as compared to the background neighboringstation audio. This may apply to inter station transition mode balance.

The background audio path can be that of analog and/or digitaldemodulated audio or bandwidth increase to the main audio source if itis digital to ensure that normalization is achieved in anticipation thatthe user will traverse or skip to it. This volume level metric alongwith the a priori information based on digital synchronization cuesenables tagging of the station as analog and digital, and may be used todecide whether the volume step increment should be triggeredautomatically if the current station is analog or if the currently tunedaudio is a digital source to employ bandwidth extension algorithms forany existing currently tuned audio source. This a priori information forthis decision can be realized by the radio head unit.

In North America, with HD IBOC there is the special situation where ananalog station on center frequency can have a digital equivalent (MainProgram Station). As such, in test drives in the field there may betransitions from analog to digital and from digital to analog. Forexample, if there is a transition from analog to digital in the case ofHD IBOC radio, the radio head unit may be able to send the HDsynchronization cue for it to know it is a transition from analog todigital and vice versa. This information may be needed in case a userskips from the current station to another station.

The above-disclosed process may be possible due to four reasons. First,Silicon On Chip (SOC) with multi-core processors supporting additionalprocessing power are providing enabling technology to signal processingalgorithms that previously was not possible to implement.

Second, software radio ASICS and SOC designs have increased processingpower to support simultaneous audio demodulation for the currently tunedstation and background stations, enabling the present invention.

Third, software radio front ends are becoming multiband and wideband aswell. What the latter means is that the front end ADC (Analog toDigital) converter can sample the entire frequency band and, dependingon the processing capability of the multicore processors, backgroundstations with potential for simultaneous audio demodulation for selectstations may be sampled. This technology can especially change thecapabilities of single tuner radios without the need for backgroundtuners for background scanning functionality.

Fourth, connectivity may also enable the option for onboard partitioninginstead of offboard partitioning for processing power needs. Thus, theradio head unit can include a single tuner and have the algorithm togauge the current audio source. The back end can perform processing forwhat used to be dedicated second or third tuner processing on the radiohead unit side through an off board server keeping track of the GPSlocation and the potential “local” frequency landscape that the car islocated in. The back end may also decode the audio for other stations inthe vicinity using an off board server to gauge the volume metric.

The invention provides a practical solution to address the use casesdefined in the table of FIGS. 5A-B. The invention may be applied to dualand multi tuner environments as well as to single tuner radios.

Use Case with Dedicated on Board Dual and/or Multi Tuner on the RadioHead Unit

Dual or Multi Tuners enable the radio to be tuned to a particularstation while the background tuners scan the other frequencies in theband. In the past, the second tuner enabled the background tuner only todo background scanning of station frequencies and did not provide audiodecode capability due to a shortage in processing power of either theSOC (System On Chip) or the ASIC (Application Specific IC). The currentstate of the art, however, includes the capability to do audiodemodulation on both the main tuner and background tuners.

With the second tuner performing audio demodulation, it is possible tocompare the perceived volume level of the currently tuned station tothat of background stations and enable automatic volume knob changesusing a pre-calibrated lookup table that associates the volume step withthe difference between the perceived volume levels of the currentlytuned station and the background stations. This automatic volume changecan be either an increase or decrease in volume levels depending on theend user's original volume level setting. This automatic volume changecan be performed for analog stations. If stations are digital, the apriori information can be used to ensure bandwidth normalization of thecurrent tuned station using the bandwidth of neighboring stations.

As an example, sampled audio of the analog demodulated audio frombackground tuners may be decomposed into N number of subbands using PQMF(Psuedo Quadrature Mirror Filters). The value N depends on the bandbeing used. In the case of analog FM where the bandwidth is capped at 15kHz, N is 32.

In the case of FM, this can be computed using:P(F)=21.4 log₁₀(4.37F+1)  (8)

Accordingly, if there is a 15 kHz sampling rate, setting F as 7.5 kHzresults in thirty-two auditory filters. For example, using the table ofFIGS. 2A-B as gauge, for AM HD where the total bandwidth can be 15 kHzthere may be thirty-two subbands.

In all situations, the common metric of the formula below may be used.The total loudness metric as applied to demodulated FM audio for aparticular volume step may thus be defined as:

$\begin{matrix}{{L = {\int_{0}^{P}{{L(p)}{dp}}}},{{{where}\mspace{14mu} p} = {32\mspace{14mu}{for}\mspace{14mu}{{FM}.}}}} & (9)\end{matrix}$

FIG. 6 shows a 512 sample audio signal frame normalized spectrum plot.FIG. 6 is a Welch power spectral density estimate as a plot ofpower/frequency (dB/Hz) vs. normalized frequency: f/(Fs/2). The darkestline represents the filtered response of the masking threshold level.The dashed line represents the masking threshold. The darkest line canin turn be calibrated to take into account the car cabin environmentnoise through a microphone input. The lighter solid line represents thatsignal for this particular 512 sample frame, which is audible to the enduser and is above the darkest line.

The present invention may make use of the (ISO/CEI norm 11172-3:1993 F)MPEG1 psychoacoustic model in Matlab. MPEG 1 may use a 512 sample framefrom an input signal, and may use the psychoacoustic model to compute aglobal masking threshold by computing the individual tonal and noisemasking frequencies utilizing the pre- and post-masking nature fortemporal effects and combining them into a final auditory maskingthreshold for that frame per subband. The algorithm may be applied toeach of the thirty-two sub bands and may return twenty-sevensignal-to-mask ratios (SMR) in dB scale, and SMR 28 to 32 are not used.For every 512 sample frame, demodulated audio levels which have signalenergy above the masking threshold may be computed for that specificsubband, and an average energy may be computed temporally across timefor the duration that the background tuner is harboring on the station.The same may be done for the main tuned station.

Since the broadcast audio is continuously changing depending on the songthat the artist is playing, the background tuner may sample stations atregular intervals and compute the appropriate energy levels above thepsychoacoustic masking level, which is what is audible by the human earfor perceived volume, and may thus define a perceived volume level. Forexample, a total loudness metric as applied to demodulated FM analogaudio for a particular volume step may be thus defined as:

$\begin{matrix}{{L = {\int_{0}^{P}{{L(p)}{dp}}}},{{{where}\mspace{14mu} p} = {32\mspace{14mu}{for}\mspace{20mu}{FM}\mspace{14mu}{for}\mspace{14mu}{{example}.}}}} & (10)\end{matrix}$

This value may be stored in RAM memory area in a table (FIG. 7) whichmay be created to map out the appropriate volume level settings thatneed to be applied to ensure that the end user perceives a specifiedvolume level. The trust timer may be used to ensure that if a stationhas been sampled, then the background tuners can scan other frequenciesin the neighboring area for the duration of the countdown of the trusttimer.

The perceived volume level comparison may take into account thefollowing three factors. The first factor is signal energy levels aboveaudio masking levels. To ensure that the algorithm scales for scenariosin the car cabin environment, microphone input to the car radio headunit can be used to ensure sampling of the background noise (e.g.,engine noise and road noise) and factoring in of the masking thresholdlevel shifts.

The second factor is the averaged audio bandwidth of the currently tunedstation vs. that of secondary neighboring stations. This can bedetermined by doing a Fast Fourier Transform of the demodulated audiofor digital and analog audio or by utilizing the modulation level ofdemodulated audio for analog audio.

The third factor is quality metrics such as fieldstrength, multipath,adjacent energy and frequency offset in the case of analog FM;fieldstrength, adjacent energy and frequency offset in the case of AM;and BER in the case of DAB signals to ensure that the algorithm does notengage when dealing with weak stations having noise which should not beamplified.

Use Case with Analog AM/FM Single Tuner without Wideband ScanningSupport

As mentioned above, the present invention may be applied to analog AM/FMsingle tuner radios as well as to dual and multi tuner radios. FIG. 8 isa table similar to the table of FIG. 7, but applicable to analogsignals. If the radio is tuned to a low modulation station on a singletuner radio, then on each audio pause the front end DSP may perform aquick alternate frequency check on two neighbor frequencies of the lowmodulation station. The alternative frequency checks occur within sevenmillisecond intervals and cannot be perceived by the user as thefrequency checks are triggered when the present audio tuned station getsa pause. Each frequency quality check may yield the fieldstrength andmodulation levels. The fieldstrength (dBuV) may be needed as themodulation level check is only valid for strong quality stationfrequencies which are characterized by a strong fieldstrength level.

The radio receiver can calculate the average modulation level of theneighboring frequencies and compare it against the modulation level ofthe currently tuned station frequency. The radio receiver can thendetermine whether the user will perceive a volume difference in theevent that the user does a tune or seek to the neighboring frequencystation. In such a circumstance, the present invention provides amechanism whereby after the tune or seek settles on the highermodulation level, the radio receiver automatically adjusts the volumelevel without user intervention.

On Multimedia Sources

DAB and HD IBOC digital audio sources are in essence similar tomultimedia audio files. In the case of a multimedia source, the sameissue arises when a user traverses from different compression rate audiofiles. Herein a background instance compressed audio decoder can be usedto gauge the volume and ensure mode balance for the current multimediasource.

While this invention has been described as having an exemplary design,the present invention may be further modified within the spirit andscope of this disclosure. This application is therefore intended tocover any variations, uses, or adaptations of the invention using itsgeneral principles. Further, this application is intended to cover suchdepartures from the present disclosure as come within known or customarypractice in the art to which this invention pertains.

What is claimed is:
 1. A radio, comprising: a first tuner and a secondtuner; and a processor configured to: compare a first volume level of astation tuned by the first tuner to at least one second volume level ofat least one background station tuned by the second tuner; and enableautomatic volume knob changes using a pre-calibrated lookup table thatassociates a volume step of the volume knob with a difference betweenthe first volume level and the second volume level.
 2. The radio ofclaim 1, wherein the automatic volume knob changes comprise volumeincreases or volume decreases depending upon an existing volume level ofthe radio.
 3. The radio of claim 1, wherein the station tuned by thefirst tuner is an analog station.
 4. The radio of claim 1, wherein thebackground station is an analog station.
 5. The radio of claim 1,comprising temporally computing an average energy of the station tunedby the first tuner, the first volume level of the station tuned by thefirst tuner being determined dependent upon the temporally computedaverage energy.
 6. The radio of claim 1, comprising temporally computingan average energy of the background station, the second volume level ofthe background station being determined dependent upon the temporallycomputed average energy.
 7. The radio of claim 1, wherein the processoris configured to implement the automatic volume knob changes inassociation with switching from audibly playing the station tuned by thefirst tuner to audibly playing the background station.
 8. A radio,comprising: a first tuner and a second tuner; and a processor configuredto: compare a first volume level of a station tuned by the first tunerto at least one second volume level of at least one background stationtuned by the second tuner; perform a volume change dependent upon adifference between the first volume level and the second volume level;and temporally compute an average energy of the station tuned by thefirst tuner, the first volume level of the station tuned by the firsttuner being determined dependent upon the temporally computed averageenergy.
 9. The radio of claim 8, wherein the processor is configured toenable automatic volume knob changes using a pre-calibrated lookup tablethat associates a volume step of the volume knob with a differencebetween the first volume level and the second volume level.
 10. Theradio of claim 8, wherein the station tuned by the first tuner is ananalog station.
 11. The radio of claim 8, wherein the background stationis an analog station.
 12. The radio of claim 8, wherein the first volumelevel comprises a first perceivable volume level.
 13. The radio of claim8, wherein the processor is configured to temporally compute an averageenergy of the background station, the second volume level of thebackground station being determined dependent upon the temporallycomputed average energy.
 14. The radio of claim 8, wherein the secondvolume level comprises a second perceivable volume level.
 15. A radio,comprising: a first tuner and a second tuner; and a processor configuredto: compare a first volume level of a station tuned by the first tunerto at least one second volume level of at least one background stationtuned by the second tuner; enable automatic volume knob changesdependent upon a volume step of the volume knob and a difference betweenthe first volume level and the second volume level; and temporallycompute an average energy of the station tuned by the first tuner, thefirst volume level of the station tuned by the first tuner beingdetermined dependent upon the temporally computed average energy. 16.The radio of claim 15, wherein the automatic volume knob changescomprise volume increases or volume decreases depending upon an existingvolume level of the radio.
 17. The radio of claim 15, wherein thestation tuned by the first tuner and/or the background station is ananalog station.
 18. The radio of claim 15, wherein the processor isconfigured to temporally compute an average energy of the backgroundstation, the second volume level of the background station beingdetermined dependent upon the temporally computed average energy. 19.The radio of claim 15, wherein at least one of the first volume leveland the second volume level is perceivable.
 20. The radio of claim 15,wherein the processor is configured to implement the automatic volumeknob changes in association with switching from audibly playing thestation tuned by the first tuner to audibly playing the backgroundstation.